Upload
posy-rodgers
View
223
Download
1
Tags:
Embed Size (px)
Citation preview
Transport Layer 3-1
Chapter 3: Transport Layer
our goals:
• understand principles behind transport layer services:– multiplexing,
demultiplexing– reliable data
transfer– flow control– congestion control
• learn about Internet transport layer protocols:– UDP: connectionless
transport– TCP: connection-
oriented reliable transport
– TCP congestion control
Transport Layer 3-2
Chapter 3 outline
3.1 transport-layer services
3.2 multiplexing and demultiplexing
3.3 connectionless transport: UDP
3.4 principles of reliable data transfer
3.5 connection-oriented transport: TCP segment structure reliable data transfer flow control connection
management
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-3
Transport services and protocols
provide logical communication between app processes running on different hosts
transport protocols run in end systems send side: breaks app
messages into segments, passes to network layer
rcv side: reassembles segments into messages, passes to app layer
more than one transport protocol available to apps Internet: TCP and UDP
applicationtransportnetworkdata linkphysical
logical end-end transport
applicationtransportnetworkdata linkphysical
Transport Layer 3-4
Transport vs. network layer
network layer: logical communication between hosts
transport layer: logical communication between processes relies on,
enhances, network layer services
12 kids in Ann’s house sending letters to 12 kids in Bill’s house:
• hosts = houses• processes = kids• app messages =
letters in envelopes• transport protocol =
Ann and Bill who demux to in-house siblings
• network-layer protocol = postal service
household analogy:
Transport Layer 3-5
Internet transport-layer protocols
• reliable, in-order delivery (TCP)– congestion control – flow control– connection setup
• unreliable, unordered delivery: UDP– no-frills extension of
“best-effort” IP
• services not available: – delay guarantees– bandwidth guarantees
applicationtransportnetworkdata linkphysical
applicationtransportnetworkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
logical end-end transport
Transport Layer 3-6
Chapter 3 outline
3.1 transport-layer services
3.2 multiplexing and demultiplexing
3.3 connectionless transport: UDP
3.4 principles of reliable data transfer
3.5 connection-oriented transport: TCP segment structure reliable data transfer flow control connection
management
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-7
Multiplexing/demultiplexing
process
socket
use header info to deliverreceived segments to correct socket
demultiplexing at receiver:handle data from multiplesockets, add transport header (later used for demultiplexing)
multiplexing at sender:
transport
application
physical
link
network
P2P1
transport
application
physical
link
network
P4
transport
application
physical
link
network
P3
Transport Layer 3-8
How demultiplexing workshost receives IP datagrams
each datagram has source IP address, destination IP address
each datagram carries one transport-layer segment
each segment has source, destination port number
host uses IP addresses & port numbers to direct segment to appropriate socket
source port # dest port #
32 bits
applicationdata (payload)
other header fields
TCP/UDP segment format
Transport Layer 3-9
Connectionless demultiplexing
• recall: created socket has host-local port #:
DatagramSocket mySocket1 = new DatagramSocket(12534);
when host receives UDP segment: checks destination port
# in segment directs UDP segment
to socket with that port #
recall: when creating datagram to send into UDP socket, must specify
destination IP address destination port #
IP datagrams with same dest. port #, but different source IP addresses and/or source port numbers will be directed to same socket at dest
Transport Layer 3-10
Connectionless demux: exampleDatagramSocket serverSocket = new DatagramSocket (6428);
transport
application
physical
link
network
P3transport
application
physical
link
network
P1
transport
application
physical
link
network
P4
DatagramSocket mySocket1 = new DatagramSocket (5775);
DatagramSocket mySocket2 = new DatagramSocket (9157);
source port: 9157dest port: 6428
source port: 6428dest port: 9157
source port: ?dest port: ?
source port: ?dest port: ?
Transport Layer 3-11
Connection-oriented demuxTCP socket
identified by 4-tuple: source IP address source port number dest IP address dest port number
demux: receiver uses all four values to direct segment to appropriate socket
server host may support many simultaneous TCP sockets: each socket identified
by its own 4-tuple
web servers have different sockets for each connecting client non-persistent HTTP
will have different socket for each request
Transport Layer 3-12
Connection-oriented demux: example
transport
application
physical
link
network
P3transport
application
physical
link
P4
transport
application
physical
link
network
P2
source IP,port: A,9157dest IP, port: B,80
source IP,port: B,80dest IP,port: A,9157
host: IP address A
host: IP address C
network
P6P5P3
source IP,port: C,5775dest IP,port: B,80
source IP,port: C,9157dest IP,port: B,80
three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets
server: IP address B
Transport Layer 3-13
Connection-oriented demux: example
transport
application
physical
link
network
P3transport
application
physical
link
transport
application
physical
link
network
P2
source IP,port: A,9157dest IP, port: B,80
source IP,port: B,80dest IP,port: A,9157
host: IP address A
host: IP address C
server: IP address B
network
P3
source IP,port: C,5775dest IP,port: B,80
source IP,port: C,9157dest IP,port: B,80
P4
threaded server
Outline
• Transport-layer services
• Multiplexing and demultiplexing
• Connectionless transport: UDP
• Principles of reliable data transfer
UDP: User Datagram Protocol [RFC 768]
• “no frills,” “bare bones” Internet transport protocol
• “best effort” service, UDP segments may be:– lost– delivered out of order to
app
• connectionless:– no handshaking
between UDP sender, receiver
– each UDP segment handled independently of others
Why is there a UDP?
• no connection establishment (which can add delay)
• simple: no connection state at sender, receiver
• small segment header
• no congestion control: UDP can blast away as fast as desired
UDP: more
• often used for streaming multimedia apps– loss tolerant– rate sensitive
• reliable transfer over UDP: add reliability at application layer– application-
specific error recovery!
source port # dest port #
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength, in
bytes of UDPsegment,including
header
Checksum
Sender:
• treat segment contents as sequence of 16-bit integers
• checksum: addition (1’s complement sum) of segment contents
• sender puts checksum value into UDP checksum field
Receiver:
• addition of all segment contents + checksum
• check if all bits are 1:– NO - error detected– YES - no error detected.
But maybe errors nonetheless? More later ….
Goal: detect “errors” (e.g., flipped bits) in transmitted segment
0110010110110100
Addition:1’s complement sum:
0110010101001111
1’s complement sum:Addition:
Transport Layer 3-18
Internet Checksum Example• Note
– When adding numbers, a carryout from the most significant bit needs to be added to the result
• Example: add two 16-bit integers
1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sumchecksum
Outline
• Reliable transfer protocols– rdt1.0: reliable transfer over a reliable channel– rdt2.0: channel with bit errors– rdt2.1: sender, handles garbled ACK/NAKs– rdt2.2: a NAK-free protocol– rdt3.0: channels with errors and loss– Pipelined protocols
• Go-back-N
• Selective repeat
• Connection-oriented transport: TCP– Overview and segment structure
Principles of Reliable data transfer• important in app., transport, link layers
• top-10 list of important networking topics!
• characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Principles of Reliable data transfer• important in app., transport, link layers
• top-10 list of important networking topics!
• characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Principles of Reliable data transfer• important in app., transport, link layers
• top-10 list of important networking topics!
• characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Reliable data transfer: getting started
sendside
receiveside
rdt_send(): called from above, (e.g., by app.). Passed data to deliver to receiver upper layer
udt_send(): called by rdt,to transfer packet over unreliable channel to
receiver
rdt_rcv(): called when packet arrives on rcv-side of channel
deliver_data(): called by rdt to deliver data to
upper
Reliable data transfer: getting started
We’ll:
• incrementally develop sender, receiver sides of reliable data transfer protocol (rdt)
• consider only unidirectional data transfer– but control info will flow on both directions!
• use finite state machines (FSM) to specify sender, receiver
state1
state2
event causing state transitionactions taken on state transition
state: when in this “state” next state
uniquely determined by next event
eventactions
Rdt1.0: reliable transfer over a reliable channel
• underlying channel perfectly reliable– no bit errors– no loss of packets
• separate FSMs for sender, receiver:– sender sends data into underlying channel– receiver read data from underlying channel
Wait for call from above packet = make_pkt(data)
udt_send(packet)
rdt_send(data)extract (packet,data)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
Rdt2.0: channel with bit errors• underlying channel may flip bits in packet
– recall: UDP checksum to detect bit errors
• the question: how to recover from errors:– acknowledgements (ACKs): receiver explicitly tells sender
that pkt received OK– negative acknowledgements (NAKs): receiver explicitly
tells sender that pkt had errors– sender retransmits pkt on receipt of NAK
• new mechanisms in rdt2.0 (beyond rdt1.0):– error detection– receiver feedback: control msgs (ACK,NAK) rcvr->sender
rdt2.0: FSM specification
Wait for call from above
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or NAK
Wait for call from below
sender
receiverrdt_send(data)
rdt2.0: operation with no errors
Wait for call from above
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or NAK
Wait for call from below
rdt_send(data)
rdt2.0: error scenario
Wait for call from above
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or NAK
Wait for call from below
rdt_send(data)
rdt2.0: FSM specification
Wait for call from above
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or NAK
Wait for call from below
sender
receiverrdt_send(data)
rdt2.0 has a fatal flaw!
What happens if ACK/NAK corrupted?
• sender doesn’t know what happened at receiver!
• can’t just retransmit: possible duplicate
What to do?
• sender ACKs/NAKs receiver’s ACK/NAK? What if sender ACK/NAK lost?
• retransmit, but this might cause retransmission of correctly received pkt!
Handling duplicates:
• sender adds sequence number to each pkt
• sender retransmits current pkt if ACK/NAK garbled
• receiver discards (doesn’t deliver up) duplicate pkt
Sender sends one packet, then waits for receiver response
stop and wait
rdt2.1: sender, handles garbled ACK/NAKs
Wait for call 0 from above
sndpkt = make_pkt(0, data, checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK
0 udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
sndpkt = make_pkt(1, data, checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt)
Wait for call 1 from
above
Wait for ACK or NAK 1
How to Draw the Receiver FSM?
• How many states does it have?
• Is it symmetric? Then focus on half
• How many events does each state have?
• What action will it be for each event?
rdt2.1: receiver, handles garbled ACK/NAKs
Wait for 0 from below
sndpkt = make_pkt(NAK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data)deliver_data(data)sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
Wait for 1 from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt)
extract(rcvpkt,data)deliver_data(data)sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
sndpkt = make_pkt(NAK, chksum)udt_send(sndpkt)
rdt2.1: discussion
Sender:
• seq # added to pkt
• two seq. #’s (0,1) will suffice. Why?
• must check if received ACK/NAK corrupted
• twice as many states– state must “remember”
whether “current” pkt has 0 or 1 seq. #
Receiver:
• must check if received packet is duplicate– state indicates whether
0 or 1 is expected pkt seq #
• Can receiver know if its last ACK/NAK received OK at sender?– No
rdt2.2: a NAK-free protocol
• same functionality as rdt2.1, using ACKs only
• instead of NAK, receiver sends ACK for last pkt received OK– receiver must explicitly include seq # of pkt being ACKed
• duplicate ACK at sender results in same action as NAK: retransmit current pkt
rdt2.2: sender, receiver fragments
Wait for call 0 from above
sndpkt = make_pkt(0, data, checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) )
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0)
Wait for ACK0
sender FSMfragment
Wait for 0 from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data)deliver_data(data)sndpkt = make_pkt(ACK1, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Outline
• Reliable transfer protocols– rdt2.1: sender, handles garbled ACK/NAKs– rdt2.2: a NAK-free protocol– rdt3.0: channels with errors and loss– Pipelined protocols
• Go-back-N
• Selective repeat
• Connection-oriented transport: TCP– Overview and segment structure
rdt3.0: channels with errors and loss
New assumption: underlying channel can also lose packets (data or ACKs)– checksum, seq. #,
ACKs, retransmissions will be of help, but not enough
Approach: sender waits “reasonable” amount of time for ACK
• retransmits if no ACK received in this time
• if pkt (or ACK) just delayed (not lost):– retransmission will be
duplicate, but use of seq. #’s already handles this
– receiver must specify seq # of pkt being ACKed
• requires countdown timer
rdt3.0 sender
sndpkt = make_pkt(0, data, checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for ACK0
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||isACK(rcvpkt,1) )
Wait for call 1 from
above
sndpkt = make_pkt(1, data, checksum)udt_send(sndpkt)start_timer
rdt_send(data)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||isACK(rcvpkt,0) )
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1)
stop_timer
stop_timer
udt_send(sndpkt)start_timer
timeout
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
Wait for ACK1
rdt_rcv(rcvpkt)
• Why is there no action taken when received packet is corrupted or ack previous packet?• What packets can be received at “wait for call 1 from above”?
rdt3.0 in action
rdt3.0 in action
Performance of rdt3.0
• rdt3.0 works, but performance stinks
• example: 1 Gbps link, 15 ms e-e prop. delay, 1KB packet:
Ttransmit
= 8kb/pkt10**9 b/sec
= 8 microsec
– U sender: utilization – fraction of time sender busy sending
U sender
= .008
30.008 = 0.00027
microseconds
L / R
RTT + L / R =
L (packet length in bits)R (transmission rate, bps)
=
– 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link– network protocol limits use of physical resources!
rdt3.0: stop-and-wait operation
first packet bit transmitted, t = 0
sender receiver
RTT
last packet bit transmitted, t = L / R
first packet bit arriveslast packet bit arrives, send ACK
ACK arrives, send next packet, t = RTT + L / R
U sender
= .008
30.008 = 0.00027
microseconds
L / R
RTT + L / R =
Outline
• Reliable transfer protocols– rdt2.1: sender, handles garbled ACK/NAKs– rdt2.2: a NAK-free protocol– rdt3.0: channels with errors and loss– Pipelined protocols
• Go-back-N
• Selective repeat
• Connection-oriented transport: TCP– Overview and segment structure
Pipelined protocolsPipelining: sender allows multiple, “in-flight”,
yet-to-be-acknowledged pkts– range of sequence numbers must be increased– buffering at sender and/or receiver
• Two generic forms of pipelined protocols: go-Back-N, selective repeat
Pipelining: increased utilization
first packet bit transmitted, t = 0
sender receiver
RTT
last bit transmitted, t = L / R
first packet bit arriveslast packet bit arrives, send ACK
ACK arrives, send next packet, t = RTT + L / R
last bit of 2nd packet arrives, send ACKlast bit of 3rd packet arrives, send ACK
U sender
= .024
30.008 = 0.0008
microseconds
3 * L / R
RTT + L / R =
Increase utilizationby a factor of 3!
Go-Back-NSender:
• k-bit seq # in pkt header
• “window” of up to N, consecutive unack’ed pkts allowed
• ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”– may deceive duplicate ACKs (see receiver)
• Single timer for all in-flight pkts
• timeout(n): retransmit pkt n and all higher seq # pkts in window
• How many states will the FSM have?
GBN: sender extended FSM
Waitstart_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])…udt_send(sndpkt[nextseqnum-1])
timeout
rdt_send(data)
if (nextseqnum < base+N) { sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ }else refuse_data(data)
base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
GBN: receiver extended FSM
ACK-only: always send ACK for correctly-received pkt with highest in-order seq #– may generate duplicate ACKs– need only remember expectedseqnum
• out-of-order pkt: – discard (don’t buffer) -> no receiver buffering!– Re-ACK pkt with highest in-order seq #
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)deliver_data(data)sndpkt = make_pkt(expectedseqnum,ACK,chksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(expectedseqnum,ACK,chksum)
GBN inaction
Selective Repeat
• receiver individually acknowledges all correctly received pkts– buffers pkts, as needed, for eventual in-order delivery
to upper layer
• sender only resends pkts for which ACK not received– sender timer for each unACKed pkt
• sender window– N consecutive seq #’s– again limits seq #s of sent, unACKed pkts
Selective repeat: sender, receiver windows
• What happened to the first two yellow packets?• There is also a mistake here … (group discussion)
Selective repeat
data from above :
• if next available seq # in window, send pkt
timeout(n):
• resend pkt n, restart timer
ACK(n) in [sendbase,sendbase+N]:
• mark pkt n as received
• if n smallest unACKed pkt, advance window base to next unACKed seq #
senderpkt n in [rcvbase, rcvbase+N-1]
• send ACK(n)
• out-of-order: buffer
• in-order: deliver (also deliver buffered, in-order pkts), advance window to next not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
• ACK(n) (earlier ack lost)
otherwise:
• ignore
receiver
Selective repeat in action
Selective repeat: dilemma
Example: seq #’s: 0, 1, 2, 3
• window size=3
• receiver sees no difference in two scenarios!
• incorrectly passes duplicate data as new in (a)
• Q: How to fix this?
• Q: what relationship between seq # size and window size?
• What about GBN ?
Outline
• Reliable transfer protocols– Pipelined protocols
• Go back N
• Selective repeat
• Connection-oriented transport: TCP– Overview and segment structure– Reliable data transfer– Flow control– Connection management
TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581
• full duplex data:– bi-directional data flow
in same connection– MSS: maximum
segment size
• connection-oriented: – handshaking
(exchange of control msgs) init’s sender, receiver state before data exchange
• flow controlled:– sender will not
overwhelm receiver
• point-to-point:– one sender, one
receiver
• reliable, in-order byte steam:– no “message
boundaries”
• pipelined:– TCP congestion and flow
control set window size
• send & receive bufferssocketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port # dest port #
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG: urgent data (generally not used)
ACK: ACK #valid
PSH: push data now(generally not used)
RST, SYN, FIN:connection estab(setup, teardown
commands)
# bytes rcvr willingto accept
countingby bytes of data(not segments!)
Internetchecksum
(as in UDP)
Compared w/ the header of UDP, what is missing?
Transport Layer 3-60
TCP seq. numbers, ACKs
sequence numbers:–byte stream “number” of first byte in segment’s data
acknowledgements:–seq # of next byte expected from other side
–cumulative ACK
Q: how receiver handles out-of-order segments–A: TCP spec doesn’t say, - up to implementor
source port # dest port #
sequence number
acknowledgement number
checksum
rwnd
urg pointer
incoming segment to sender
A
sent ACKed
sent, not-yet ACKed(“in-flight”)
usablebut not yet sent
not usable
window size N
sender sequence number space
source port # dest port #
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-61
TCP seq. numbers, ACKs
Usertypes
‘C’
host ACKsreceipt
of echoed‘C’
host ACKsreceipt of‘C’, echoesback ‘C’
simple telnet scenario
Host BHost A
Seq=42, ACK=79, data = ‘C’
Seq=79, ACK=43, data = ‘C’
Seq=43, ACK=80
TCP Round Trip Time and Timeout
Q: how to set TCP timeout value?
• longer than RTT– but RTT varies
• too short: premature timeout– unnecessary
retransmissions
• too long: slow reaction to segment loss
Q: how to estimate RTT?
• SampleRTT: measured time from segment transmission until ACK receipt– ignore retransmissions
• SampleRTT will vary, want estimated RTT “smoother”– average several recent
measurements, not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
• Exponential weighted moving average
• influence of past sample decreases exponentially fast
• typical value: = 0.125
Example RTT estimation:RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout• EstimtedRTT plus “safety margin”
– large variation in EstimatedRTT -> larger safety margin
• first estimate of how much SampleRTT deviates from EstimatedRTT:
DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|
(typically, = 0.25)
Then set timeout interval:
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT “safety margin”
Outline
• Reliable transfer protocols– Pipelined protocols
• Selective repeat
• Connection-oriented transport: TCP– Overview and segment structure– Reliable data transfer– Flow control– Connection management
TCP reliable data transfer
• TCP creates rdt service on top of IP’s unreliable service
• Pipelined segments
• Cumulative acks
• TCP uses single retransmission timer
• Retransmissions are triggered by:– timeout events– duplicate acks
• Initially consider simplified TCP sender:– ignore duplicate acks– ignore flow control,
congestion control
TCP sender events:data rcvd from app:
• Create segment with seq #
• seq # is byte-stream number of first data byte in segment
• start timer if not already running (think of timer as for oldest unacked segment)
• expiration interval: TimeOutInterval
timeout:
• retransmit segment that caused timeout
• restart timer
Ack rcvd:
• If acknowledges previously unacked segments– update what is known to be
acked– start timer if there are
outstanding segments
• Difference from GBN?
Transport Layer 3-69
TCP sender (simplified)
waitfor event
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
create segment, seq. #: NextSeqNumpass segment to IP (i.e., “send”)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer
data received from application above
retransmit not-yet-acked segment with smallest seq. #start timer
timeout
if (y > SendBase) { SendBase = y /* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments) start timer else stop timer }
ACK received, with ACK field value y
TCP: retransmission scenarios
Host A
Seq=100, 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92, 8 bytes data
ACK=120
Seq=92, 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92, 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92, 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92, 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100, 20 bytes data
ACK=120
time
SendBase= 120
TCP ACK generation [RFC 1122, RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq #. All data up toexpected seq # already ACKed
Arrival of in-order segment withexpected seq #. One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq. # .Gap detected
Arrival of segment that partially or completely fills gap
TCP Receiver action
Delayed ACK. Wait up to 500msfor next segment. If no next segment,send ACK
Immediately send single cumulative ACK, ACKing both in-order segments
Immediately send duplicate ACK, indicating seq. # of next expected byte
Immediate send ACK, provided thatsegment starts at lower end of gap
Fast Retransmit
• Time-out period often relatively long:– long delay before
resending lost packet
• Detect lost segments via duplicate ACKs.– Sender often sends
many segments back-to-back
– If segment is lost, there will likely be many duplicate ACKs.
• If sender receives 3 ACKs for the same data, it supposes that segment after ACKed data was lost:– fast retransmit: resend
segment before timer expires
event: ACK received, with ACK field value of y if (y > SendBase) { SendBase = y if (there are currently not-yet-acknowledged segments) start timer } else { increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) { resend segment with sequence number y }
Fast retransmit algorithm:
a duplicate ACK for already ACKed segment
fast retransmit
Outline
• Flow control
• Connection management
• Congestion control
TCP Flow Control
• receive side of TCP connection has a receive buffer:
• speed-matching service: matching the send rate to the receiving app’s drain rate
• app process may be slow at reading from buffer
sender won’t overflow
receiver’s buffer bytransmitting too
much, too fast
flow control
TCP Flow control: how it works
(Suppose TCP receiver discards out-of-order segments)
• spare room in buffer
= RcvWindow
= RcvBuffer-[LastByteRcvd - LastByteRead]
• Rcvr advertises spare room by including value of RcvWindow in segments
• Sender limits unACKed data to RcvWindow– guarantees receive
buffer doesn’t overflow
TCP Connection Management
Recall: TCP sender, receiver establish “connection” before exchanging data segments
• initialize TCP variables:– seq. #s– buffers, flow control
info (e.g. RcvWindow)
• client: connection initiator
• server: contacted by client
Three way handshake:
Step 1: client host sends TCP SYN segment to server– specifies initial seq #– no data
Step 2: server host receives SYN, replies with SYNACK segment
– server allocates buffers– specifies server initial seq.
#
Step 3: client receives SYNACK, replies with ACK segment, which may contain data
What about if the client does not send SYN ACK …
TCP Connection Management: Closing
Step 1: client end system sends TCP FIN control segment to server
Step 2: server receives FIN, replies with ACK. Closes connection, sends FIN.
Step 3: client receives FIN, replies with ACK.
– Enters “timed wait” - will respond with ACK to received FINs
Step 4: server, receives ACK. Connection closed.
Note: with small modification, can handle simultaneous FINs
client
FIN
server
ACK
ACK
FIN
closing
closing
closedti
med w
ait
closed
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Outline
• Flow control
• Connection management
• Congestion control
Principles of Congestion Control
Congestion:
• informally: “too many sources sending too much data too fast for network to handle”
• different from flow control!
• manifestations:– lost packets (buffer overflow at routers)– long delays (queueing in router buffers)
• Reasons– Limited bandwidth, queues– Unneeded retransmission for data and ACKs
Approaches towards congestion control
End-end congestion control:
• no explicit feedback from network
• congestion inferred from end-system observed loss, delay
• approach taken by TCP
Network-assisted congestion control:
• routers provide feedback to end systems– single bit indicating
congestion (SNA, DECbit, TCP/IP ECN, ATM)
– explicit rate sender should send at
Two broad approaches towards congestion control:
TCP Congestion Control
• end-end control (no network assistance)
• sender limits transmission:
LastByteSent-LastByteAcked
CongWin
• Roughly,
• CongWin is dynamic, function of perceived network congestion
How does sender perceive congestion?
• loss event = timeout or 3 duplicate acks
• TCP sender reduces rate (CongWin) after loss event
three mechanisms:– AIMD– slow start– conservative after
timeout events
rate = CongWin
RTT Bytes/sec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease: cut CongWin in half after loss event
additive increase: increase CongWin by 1 MSS every RTT in the absence of loss events: probing
Long-lived TCP connection
TCP Slow Start
• When connection begins, CongWin = 1 MSS– Example: MSS = 500
bytes & RTT = 200 msec
– initial rate = 20 kbps
• available bandwidth may be >> MSS/RTT– desirable to quickly
ramp up to respectable rate
• When connection begins, increase rate exponentially fast until first loss event
TCP Slow Start (more)
• When connection begins, increase rate exponentially until first loss event:– double CongWin every
RTT– done by incrementing CongWin for every ACK received
• Summary: initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement (more)Q: When should the
exponential increase switch to linear?
A: When CongWin gets to 1/2 of its value before timeout.
Implementation:
• Variable Threshold
• At loss event, Threshold is set to 1/2 of CongWin just before loss event
0
2
4
6
8
10
12
14
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
co
ng
es
tio
n w
ind
ow
siz
e
(se
gm
en
ts)
threshold
Refinement
• After 3 dup ACKs:– CongWin is cut in half– window then grows
linearly
• But after timeout event:– Enter “slow start”– CongWin instead set to
1 MSS; – window then grows
exponentially– to a threshold, then
grows linearly
• 3 dup ACKs indicates network capable of delivering some segments• timeout before 3 dup ACKs is “more alarming”
Philosophy:
Summary: TCP Congestion Control
• When CongWin is below Threshold, sender in slow-start phase, window grows exponentially.
• When CongWin is above Threshold, sender is in congestion-avoidance phase, window grows linearly.
• When a triple duplicate ACK occurs, Threshold set to CongWin/2 and CongWin set to Threshold.
• When timeout occurs, Threshold set to CongWin/2 and CongWin is set to 1 MSS.
Fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair?Two competing sessions:
• Additive increase gives slope of 1, as throughout increases
• multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 sending rateConnect
ion 2
sen
di n
g r
ate
congestion avoidance: additive increaseloss: decrease window by factor of 2
congestion avoidance: additive increaseloss: decrease window by factor of 2
Fairness (more)
Fairness and UDP
• Multimedia apps often do not use TCP– do not want rate
throttled by congestion control
• Instead use UDP:– pump audio/video at
constant rate, tolerate packet loss
• Research area: TCP friendly
Fairness and parallel TCP connections
• nothing prevents app from opening parallel connections between 2 hosts.
• Web browsers do this
• Example: link of rate R supporting 9 connections; – new app asks for 1 TCP, gets
rate R/10– new app asks for 11 TCPs,
gets R/2 !
Midterm
• Next Wed. 3:30-5pm, same classroom
• Closed book, one-page cheat sheet
• T/F, short answer and in-depth, similar to hw
• Covers Ch. 1-3 and project 1
• No count-to-infinity qn, but need to know arithmetic